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externals/openal-soft/examples/alconvolve.c

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+/*
+ * OpenAL Convolution Reverb Example
+ *
+ * Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/* This file contains an example for applying convolution reverb to a source. */
+
+#include <assert.h>
+#include <inttypes.h>
+#include <limits.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "sndfile.h"
+
+#include "AL/al.h"
+#include "AL/alext.h"
+
+#include "common/alhelpers.h"
+
+
+#ifndef AL_SOFT_convolution_reverb
+#define AL_SOFT_convolution_reverb
+#define AL_EFFECT_CONVOLUTION_REVERB_SOFT        0xA000
+#endif
+
+
+/* Filter object functions */
+static LPALGENFILTERS alGenFilters;
+static LPALDELETEFILTERS alDeleteFilters;
+static LPALISFILTER alIsFilter;
+static LPALFILTERI alFilteri;
+static LPALFILTERIV alFilteriv;
+static LPALFILTERF alFilterf;
+static LPALFILTERFV alFilterfv;
+static LPALGETFILTERI alGetFilteri;
+static LPALGETFILTERIV alGetFilteriv;
+static LPALGETFILTERF alGetFilterf;
+static LPALGETFILTERFV alGetFilterfv;
+
+/* Effect object functions */
+static LPALGENEFFECTS alGenEffects;
+static LPALDELETEEFFECTS alDeleteEffects;
+static LPALISEFFECT alIsEffect;
+static LPALEFFECTI alEffecti;
+static LPALEFFECTIV alEffectiv;
+static LPALEFFECTF alEffectf;
+static LPALEFFECTFV alEffectfv;
+static LPALGETEFFECTI alGetEffecti;
+static LPALGETEFFECTIV alGetEffectiv;
+static LPALGETEFFECTF alGetEffectf;
+static LPALGETEFFECTFV alGetEffectfv;
+
+/* Auxiliary Effect Slot object functions */
+static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
+static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
+static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
+static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
+static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
+static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
+static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
+static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
+static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
+static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
+static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
+
+
+/* This stuff defines a simple streaming player object, the same as alstream.c.
+ * Comments are removed for brevity, see alstream.c for more details.
+ */
+#define NUM_BUFFERS 4
+#define BUFFER_SAMPLES 8192
+
+typedef struct StreamPlayer {
+    ALuint buffers[NUM_BUFFERS];
+    ALuint source;
+
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    float *membuf;
+
+    ALenum format;
+} StreamPlayer;
+
+static StreamPlayer *NewPlayer(void)
+{
+    StreamPlayer *player;
+
+    player = calloc(1, sizeof(*player));
+    assert(player != NULL);
+
+    alGenBuffers(NUM_BUFFERS, player->buffers);
+    assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
+
+    alGenSources(1, &player->source);
+    assert(alGetError() == AL_NO_ERROR && "Could not create source");
+
+    alSource3i(player->source, AL_POSITION, 0, 0, -1);
+    alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
+    alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
+    assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
+
+    return player;
+}
+
+static void ClosePlayerFile(StreamPlayer *player)
+{
+    if(player->sndfile)
+        sf_close(player->sndfile);
+    player->sndfile = NULL;
+
+    free(player->membuf);
+    player->membuf = NULL;
+}
+
+static void DeletePlayer(StreamPlayer *player)
+{
+    ClosePlayerFile(player);
+
+    alDeleteSources(1, &player->source);
+    alDeleteBuffers(NUM_BUFFERS, player->buffers);
+    if(alGetError() != AL_NO_ERROR)
+        fprintf(stderr, "Failed to delete object IDs\n");
+
+    memset(player, 0, sizeof(*player));
+    free(player);
+}
+
+static int OpenPlayerFile(StreamPlayer *player, const char *filename)
+{
+    size_t frame_size;
+
+    ClosePlayerFile(player);
+
+    player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
+    if(!player->sndfile)
+    {
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
+        return 0;
+    }
+
+    player->format = AL_NONE;
+    if(player->sfinfo.channels == 1)
+        player->format = AL_FORMAT_MONO_FLOAT32;
+    else if(player->sfinfo.channels == 2)
+        player->format = AL_FORMAT_STEREO_FLOAT32;
+    else if(player->sfinfo.channels == 6)
+        player->format = AL_FORMAT_51CHN32;
+    else if(player->sfinfo.channels == 3)
+    {
+        if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
+            player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
+    }
+    else if(player->sfinfo.channels == 4)
+    {
+        if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
+            player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
+    }
+    if(!player->format)
+    {
+        fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
+        sf_close(player->sndfile);
+        player->sndfile = NULL;
+        return 0;
+    }
+
+    frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
+    player->membuf = malloc(frame_size);
+
+    return 1;
+}
+
+static int StartPlayer(StreamPlayer *player)
+{
+    ALsizei i;
+
+    alSourceRewind(player->source);
+    alSourcei(player->source, AL_BUFFER, 0);
+
+    for(i = 0;i < NUM_BUFFERS;i++)
+    {
+        sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
+        if(slen < 1) break;
+
+        slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
+        alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
+            player->sfinfo.samplerate);
+    }
+    if(alGetError() != AL_NO_ERROR)
+    {
+        fprintf(stderr, "Error buffering for playback\n");
+        return 0;
+    }
+
+    alSourceQueueBuffers(player->source, i, player->buffers);
+    alSourcePlay(player->source);
+    if(alGetError() != AL_NO_ERROR)
+    {
+        fprintf(stderr, "Error starting playback\n");
+        return 0;
+    }
+
+    return 1;
+}
+
+static int UpdatePlayer(StreamPlayer *player)
+{
+    ALint processed, state;
+
+    alGetSourcei(player->source, AL_SOURCE_STATE, &state);
+    alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
+    if(alGetError() != AL_NO_ERROR)
+    {
+        fprintf(stderr, "Error checking source state\n");
+        return 0;
+    }
+
+    while(processed > 0)
+    {
+        ALuint bufid;
+        sf_count_t slen;
+
+        alSourceUnqueueBuffers(player->source, 1, &bufid);
+        processed--;
+
+        slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
+        if(slen > 0)
+        {
+            slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
+            alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
+                player->sfinfo.samplerate);
+            alSourceQueueBuffers(player->source, 1, &bufid);
+        }
+        if(alGetError() != AL_NO_ERROR)
+        {
+            fprintf(stderr, "Error buffering data\n");
+            return 0;
+        }
+    }
+
+    if(state != AL_PLAYING && state != AL_PAUSED)
+    {
+        ALint queued;
+
+        alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
+        if(queued == 0)
+            return 0;
+
+        alSourcePlay(player->source);
+        if(alGetError() != AL_NO_ERROR)
+        {
+            fprintf(stderr, "Error restarting playback\n");
+            return 0;
+        }
+    }
+
+    return 1;
+}
+
+
+/* CreateEffect creates a new OpenAL effect object with a convolution reverb
+ * type, and returns the new effect ID.
+ */
+static ALuint CreateEffect(void)
+{
+    ALuint effect = 0;
+    ALenum err;
+
+    printf("Using Convolution Reverb\n");
+
+    /* Create the effect object and set the convolution reverb effect type. */
+    alGenEffects(1, &effect);
+    alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
+
+    /* Check if an error occured, and clean up if so. */
+    err = alGetError();
+    if(err != AL_NO_ERROR)
+    {
+        fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
+        if(alIsEffect(effect))
+            alDeleteEffects(1, &effect);
+        return 0;
+    }
+
+    return effect;
+}
+
+/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
+ * returns the new buffer ID.
+ */
+static ALuint LoadSound(const char *filename)
+{
+    const char *namepart;
+    ALenum err, format;
+    ALuint buffer;
+    SNDFILE *sndfile;
+    SF_INFO sfinfo;
+    float *membuf;
+    sf_count_t num_frames;
+    ALsizei num_bytes;
+
+    /* Open the audio file and check that it's usable. */
+    sndfile = sf_open(filename, SFM_READ, &sfinfo);
+    if(!sndfile)
+    {
+        fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
+        return 0;
+    }
+    if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
+    {
+        fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
+        sf_close(sndfile);
+        return 0;
+    }
+
+    /* Get the sound format, and figure out the OpenAL format. Use floats since
+     * impulse responses will usually have more than 16-bit precision.
+     */
+    format = AL_NONE;
+    if(sfinfo.channels == 1)
+        format = AL_FORMAT_MONO_FLOAT32;
+    else if(sfinfo.channels == 2)
+        format = AL_FORMAT_STEREO_FLOAT32;
+    else if(sfinfo.channels == 3)
+    {
+        if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
+            format = AL_FORMAT_BFORMAT2D_FLOAT32;
+    }
+    else if(sfinfo.channels == 4)
+    {
+        if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
+            format = AL_FORMAT_BFORMAT3D_FLOAT32;
+    }
+    if(!format)
+    {
+        fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
+        sf_close(sndfile);
+        return 0;
+    }
+
+    namepart = strrchr(filename, '/');
+    if(namepart || (namepart=strrchr(filename, '\\')))
+        namepart++;
+    else
+        namepart = filename;
+    printf("Loading: %s (%s, %dhz, %" PRId64 " samples / %.2f seconds)\n", namepart,
+        FormatName(format), sfinfo.samplerate, sfinfo.frames,
+        (double)sfinfo.frames / sfinfo.samplerate);
+    fflush(stdout);
+
+    /* Decode the whole audio file to a buffer. */
+    membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
+
+    num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
+    if(num_frames < 1)
+    {
+        free(membuf);
+        sf_close(sndfile);
+        fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
+        return 0;
+    }
+    num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
+
+    /* Buffer the audio data into a new buffer object, then free the data and
+     * close the file.
+     */
+    buffer = 0;
+    alGenBuffers(1, &buffer);
+    alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
+
+    free(membuf);
+    sf_close(sndfile);
+
+    /* Check if an error occured, and clean up if so. */
+    err = alGetError();
+    if(err != AL_NO_ERROR)
+    {
+        fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
+        if(buffer && alIsBuffer(buffer))
+            alDeleteBuffers(1, &buffer);
+        return 0;
+    }
+
+    return buffer;
+}
+
+
+int main(int argc, char **argv)
+{
+    ALuint ir_buffer, filter, effect, slot;
+    StreamPlayer *player;
+    int i;
+
+    /* Print out usage if no arguments were specified */
+    if(argc < 2)
+    {
+        fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> "
+            "<[-dry | -nodry] filename>...\n", argv[0]);
+        return 1;
+    }
+
+    argv++; argc--;
+    if(InitAL(&argv, &argc) != 0)
+        return 1;
+
+    if(!alIsExtensionPresent("AL_SOFTX_convolution_reverb"))
+    {
+        CloseAL();
+        fprintf(stderr, "Error: Convolution revern not supported\n");
+        return 1;
+    }
+
+    if(argc < 2)
+    {
+        CloseAL();
+        fprintf(stderr, "Error: Missing impulse response or sound files\n");
+        return 1;
+    }
+
+    /* Define a macro to help load the function pointers. */
+#define LOAD_PROC(T, x)  ((x) = FUNCTION_CAST(T, alGetProcAddress(#x)))
+    LOAD_PROC(LPALGENFILTERS, alGenFilters);
+    LOAD_PROC(LPALDELETEFILTERS, alDeleteFilters);
+    LOAD_PROC(LPALISFILTER, alIsFilter);
+    LOAD_PROC(LPALFILTERI, alFilteri);
+    LOAD_PROC(LPALFILTERIV, alFilteriv);
+    LOAD_PROC(LPALFILTERF, alFilterf);
+    LOAD_PROC(LPALFILTERFV, alFilterfv);
+    LOAD_PROC(LPALGETFILTERI, alGetFilteri);
+    LOAD_PROC(LPALGETFILTERIV, alGetFilteriv);
+    LOAD_PROC(LPALGETFILTERF, alGetFilterf);
+    LOAD_PROC(LPALGETFILTERFV, alGetFilterfv);
+
+    LOAD_PROC(LPALGENEFFECTS, alGenEffects);
+    LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
+    LOAD_PROC(LPALISEFFECT, alIsEffect);
+    LOAD_PROC(LPALEFFECTI, alEffecti);
+    LOAD_PROC(LPALEFFECTIV, alEffectiv);
+    LOAD_PROC(LPALEFFECTF, alEffectf);
+    LOAD_PROC(LPALEFFECTFV, alEffectfv);
+    LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
+    LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
+    LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
+    LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
+
+    LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
+    LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
+    LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
+    LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
+    LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
+    LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
+    LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
+    LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
+    LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
+    LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
+    LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
+#undef LOAD_PROC
+
+    /* Load the reverb into an effect. */
+    effect = CreateEffect();
+    if(!effect)
+    {
+        CloseAL();
+        return 1;
+    }
+
+    /* Load the impulse response sound into a buffer. */
+    ir_buffer = LoadSound(argv[0]);
+    if(!ir_buffer)
+    {
+        alDeleteEffects(1, &effect);
+        CloseAL();
+        return 1;
+    }
+
+    /* Create the effect slot object. This is what "plays" an effect on sources
+     * that connect to it.
+     */
+    slot = 0;
+    alGenAuxiliaryEffectSlots(1, &slot);
+
+    /* Set the impulse response sound buffer on the effect slot. This allows
+     * effects to access it as needed. In this case, convolution reverb uses it
+     * as the filter source. NOTE: Unlike the effect object, the buffer *is*
+     * kept referenced and may not be changed or deleted as long as it's set,
+     * just like with a source. When another buffer is set, or the effect slot
+     * is deleted, the buffer reference is released.
+     *
+     * The effect slot's gain is reduced because the impulse responses I've
+     * tested with result in excessively loud reverb. Is that normal? Even with
+     * this, it seems a bit on the loud side.
+     *
+     * Also note: unlike standard or EAX reverb, there is no automatic
+     * attenuation of a source's reverb response with distance, so the reverb
+     * will remain full volume regardless of a given sound's distance from the
+     * listener. You can use a send filter to alter a given source's
+     * contribution to reverb.
+     */
+    alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
+    alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
+    alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
+    assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
+
+    /* Create a filter that can silence the dry path. */
+    filter = 0;
+    alGenFilters(1, &filter);
+    alFilteri(filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
+    alFilterf(filter, AL_LOWPASS_GAIN, 0.0f);
+
+    player = NewPlayer();
+    /* Connect the player's source to the effect slot. */
+    alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
+    assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
+
+    /* Play each file listed on the command line */
+    for(i = 1;i < argc;i++)
+    {
+        const char *namepart;
+
+        if(argc-i > 1)
+        {
+            if(strcasecmp(argv[i], "-nodry") == 0)
+            {
+                alSourcei(player->source, AL_DIRECT_FILTER, (ALint)filter);
+                ++i;
+            }
+            else if(strcasecmp(argv[i], "-dry") == 0)
+            {
+                alSourcei(player->source, AL_DIRECT_FILTER, AL_FILTER_NULL);
+                ++i;
+            }
+        }
+
+        if(!OpenPlayerFile(player, argv[i]))
+            continue;
+
+        namepart = strrchr(argv[i], '/');
+        if(namepart || (namepart=strrchr(argv[i], '\\')))
+            namepart++;
+        else
+            namepart = argv[i];
+
+        printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
+            player->sfinfo.samplerate);
+        fflush(stdout);
+
+        if(!StartPlayer(player))
+        {
+            ClosePlayerFile(player);
+            continue;
+        }
+
+        while(UpdatePlayer(player))
+            al_nssleep(10000000);
+
+        ClosePlayerFile(player);
+    }
+    printf("Done.\n");
+
+    /* All files done. Delete the player and effect resources, and close down
+     * OpenAL.
+     */
+    DeletePlayer(player);
+    player = NULL;
+
+    alDeleteAuxiliaryEffectSlots(1, &slot);
+    alDeleteEffects(1, &effect);
+    alDeleteFilters(1, &filter);
+    alDeleteBuffers(1, &ir_buffer);
+
+    CloseAL();
+
+    return 0;
+}